Basic processing of audio samples in wav format, using fourier transformation.
% wav file in PCM forma to morph
% frequency shiftiting
%frequency will be cutt off due to the window size
% make a new coefficient matrix composed by zeros
% the DC component must be preserved!
% the even frequency must be copied
% if frequencies are shifted up
if(nshift>0 && nshift<Nhalf)
newcoeff(nshift+1:Nhalf) = coeff(2:Nhalf-nshift+1);
% if frequencies are shifted down
elseif(nshift<0 && nshift>floor(-N/2))
fprintf('Frequency shift is not possible.\n')